I am trying to relay a SIP SUBSCRIBE message from OpenSIPS to my ejabberd server. However ejabberd insists on challenging this with a 407 Proxy Authentication Required response.
I would like to somehow add my OpenSIPs server as a trusted peer on ejabberd - is this possible at all? I've been playing with the acls and various parts of configuration but nothing seems to do the trick.
Pete
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I would like to set up a proxy for SMTP traffic by this i mean send and receive mail through a proxy.
I have a machine which is not directly connect to an internet gateaway, I already have an squid proxy for http and https but now i need to send mail (receive is not fundamental)
Which tools do you advice me to use and if you have some doc or tutorial to use this tool as an smtp proxy it would be great. (I'm running on CentOS 7)
Thanks in advance
This question is very specific to FRITZ!Box 7360. I have the lest firmware installed.
I want to configure a sip account from a provider like Elvero. I tried to follow instructions provided at configuring internet telephone. After configuring I do not see any sip connection attempts reaching the sip server and FRITZ!Box complains that it did not get a response from the server.
This means, I cannot receive and make calls through Elvero. Anyone got successful in connecting FRITZ!Box to an external sip provider?
I want to connect to my ejabberd server from another machine using a certificate instead of a login/password. I've looked for authentication client-to-server with a certificate for ejabberd, but i couldn't find something helpfull.
If anyone has any ideas how it cas be done, I'm taking..
As of version 16.02, ejabberd Community Server does not yet support client cert authentication.
However, if your questions is about communication encryption, you can indeed configure ejabberd with Starttls support to use TLS between client and server. A service like Let's encrypt can provide such certificates for free: https://letsencrypt.org/
I'm very very new to the world of VoIP, so please forgive my terminological inaccuracies.
I have just installed ( on to a remote server ) the OpenSIPS software.
I have provisioned a Twilio Elastic SIP Trunk which has it's Origination SIP URI set to the IP address of the OpenSIPS server ( but it prepended with the protocol 'sip:' )
I have two iPhones with some VoIP apps, that allow me to specify Auth Name, Auth Password, Username, Display Name, Domain and SIP Proxy.
In OpenSIPS i have created a domain ( with the IP of my server )
also, i created two users ( on that domain ) and gave them Aliases
of 100 and 101.
The software on the iPhones claims that it has 'registered' with the
OpenSIPS server, so I have to accept that.
I believe that my configuration of the Twilio SIP Trunk is correct, on the
basis that there aren't many settings to set, and they all look correct,
and make some kind of logical sense.
What i cant figure out from here, and not for lack of searching the web,
watching hours of webinars etc, is how do I get OpenSIPS to send outbound
calls through to the Twilio Trunk, and also get the handsets to ring on
incomming calls, and Twilio's SIP Trunk has a Termination SIP URI of xxxxxx.pstn.twilio.com which is ( im guessing ) supposed to be entered somewhere in to OpenSIPS but I cannot see how to configure this.
I'm looking for something that is a guide on how to 'connect' OpenSIPS to the Twilio Elastic Sip Trunking service.
Any help appreciated...
So, I'm guessing you want to set up an OpenSIPS instance which acts both as SIP registrar and basic SBC, by simply forwarding all outgoing traffic to the Twilio SIP trunk.
SIP Registrar
It looks like you've already handled the registrar part of OpenSIPS scripting. You can always generate a basic configuration that handles user registrations with:
make menuconfig
"Generate OpenSIPS script"
"Residential script"
To list the currently registered OpenSIPS users:
opensipsctl ul show
SIP Session Border Controller (basic routing)
- managing outgoing traffic
Here are a few useful scripting mechanisms:
$ru / $rU / $rd variables (management of Entire SIP Request URI / Just username of R-URI / Just domain of R-URI) - do all necessary changes (i.e. force the R-URI to be xxxxxx.pstn.twilio.com...)
$du variable - (management of Outbound Proxy) - this will help us route to the Twilio SIP trunk. Setting a destination URI, then doing t_relay() will simply proxy the current request to the specified destination. For a more advanced routing mechanism (i.e. you need to route to a bunch of trunks), consider using the dispatcher module, which supports using a cached SQL table populated with SIP destinations.
- managing incoming traffic
the registrar module functions should be enough for this (test with lookup() or registered(), then just do t_relay()). They will verify if your iPhones are currently registered on the OpenSIPS server.
I set up a BOSH server using node-xmpp-bosh and configured converse.js (a javascript XMPP client) to use the BOSH url I set up, I tried log in with my jabber.org account on converse.js and all worked fine.
Now I need to make this work with a private XMPP server. Assume that the XMPP server is at chat.mysite.com:5222, a JID on the server is johndoe#mysite.com. How do I configure the BOSH server so that it connects to chat.mysite.com:5222 when I try to log in with johndoe#mysite.com on converse.js?
You should add DNS SRV record _xmpp-client._tcp.mysite.com which should point to chat.mysite.com 5222